asterisk users - Feb 2012 - DTMF forwarding and Page

Hi,
I'd like to implement some way of controlling remote SIP clients while 
in a call, to execute remote commands.

The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected 
to the previous one

I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().

TIA,
Matteo

Noone knows that? Where/whom could I ask?

Thanks

Il 10/02/2012 12:30, Matteo Fortini ha scritto:
> Hi,
> I'd like to implement some way of controlling remote SIP clients while 
> in a call, to execute remote commands.
>
> The call topology (think of a PA system) is this:
> * the caller is in a MeetMe() conference room
> * the callees are Page()d, then the dynamic conference room is 
> connected to the previous one
>
> I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
> caller to the callees. I found option 'F' for MeetMe, but I have no
> control on Page().
>
> TIA,
> Matteo